In the past, two important types of communication network for transferring information have developed: packet-oriented (data) networks and line-oriented (voice) networks. The convergence of these two network types has led to the development of convergent (voice-data) networks. Merging these different network types has resulted in hybrid networks, in which the subject matter of the present invention is utilized with particularly noteworthy advantages.
Line-oriented networks—also called voice networks or telephone networks—are designed for the transfer of continuously streaming (voice) information, this being referred to as a call or session by experts. In this type of activity, the information transfer is usually characterized by a high quality of service and reliability. For example, a minimal delay—e.g. <200 ms—without delay time fluctuations (delay jitter) is important for voice, since voice requires a continuous information flow when it is reproduced in the receiving device. Therefore an information loss cannot be compensated by re-transferring the information which was not transferred, and usually results in acoustically noticeable crackling. Experts generally refer to the transfer of voice as a ‘realtime service’. transfer of packet streams, which are also referred to as ‘data packet streams’ or ‘flow’ by experts. In this type of activity, it is not usually necessary to guarantee a high quality of service. Without a guaranteed quality of service, the transfer of the data packet streams is subject to e.g. temporally fluctuating delays, since the individual data packets of the data packet streams are usually transferred in the sequence in which they enter the network, i.e. the time delays increase in accordance with the number of packets that must be transferred by a network. Therefore experts also refer to the transfer of data as a transfer service without realtime conditions or as a ‘non-realtime service’.
Depending on the type of packet-oriented network, the packets can be configured as e.g. Internet packets, X.25 packets, frame-relay packets, or even ATM cells. They are sometimes also referred to as messages, primarily if a message is transferred in a packet.
A well-known data network is the Internet. This is also called an IP network sometimes, due to the Internet protocol IP which is used there, wherein this term should generally be understood to have a broad sense and include all networks in which the IP protocol is used. The Internet is designed as an open (wide area) data network having open interfaces for connecting (mainly local and regional) data networks of different manufacturers. It provides a manufacturer-independent transport platform.
Connections are communication links between at least two subscribers for the purpose of two-way information transfer. The subscriber initiating the connection is normally called the ‘A-subscriber’. A subscriber who is connected to an A-subscriber as a result of a connection is called a ‘B-subscriber’. In a connectionless network, connections represent at least the relationship between A-subscriber and B-subscriber, said relationship being specific on a logically abstract level, i.e. the connectionless flows in the Internet, for example, represent logically abstract connections accordingly (e.g. A-subscriber=browser and B-subscriber=web server). In a connection-oriented network, connections represent paths through the network, along which paths the information is transferred, said paths also being specific on a physical level.
As a result of the convergence of voice and data networks, packet-oriented networks are likewise being used for implementing voice transfer services and increasingly also for implementing services that require more bandwidth such as e.g. transfer of moving-image information, i.e. the transfer of realtime services which previously usually involved line-oriented transfer takes place in a convergent network—also called a voice-data network—in a packet-oriented manner, i.e. in packet streams. These are also called realtime packet streams. In this case, the transfer of voice information via a packet-oriented IP network is also called ‘VoIP’ (Voice over IP).
A plurality of architectures for voice-data networks are described in the international standardization bodies IETF (International Engineering Task Force) and ITU (International Telecommunications Union). It is common to all that the Call Control layer and the Resource Control Layer are clearly separate from each other in functional terms.
In this case, the Call Control layer comprises at least one (optional) Call Controller, to which inter alia the following functions are assigned:                Address Translation: conversion of E.164 telephone numbers and other alias addresses (e.g. computer names) into transport addresses (e.g. Internet addresses).        Admission Control (optional): basic validity check for determining whether and to what extent (e.g. VoIP-compatible) entities are allowed to utilize the communication network.        Bandwidth Control (optional): management of transfer capacities.        Zone Management: registration of (e.g. VoIP-compatible) entities and provision of above functions for all entities registered at the Call Controller.        
In addition, the following functions can optionally be assigned to a Call Controller if necessary:                Call Control Signaling: all signaling messages are switched by at least one Call Controller, i.e. all entities send and receive signaling messages only via the Call Controller. Any direct exchange of signaling messages between the entities is prohibited.        Call Authorization: validity check for incoming and outgoing calls.        Bandwidth Management: controlling the permitted number of entities which are allowed to utilize the communication network concurrently.        Call Management: managing a list of current calls, e.g. so that it is possible to generate a busy tone if this cannot be generated by the entity itself.        Alias Address Modification: returning a modified Alias Address, e.g. with an H.225.0 message ACF (Admission Confirmation). The endpoint must use this address during connection setup.        Dialed Digit Translation: translating the dialed digits into an E.164 telephone number or into a number from a private numbering model.        
The ‘Gatekeeper’ proposed by the ITU in the H.323 family of standards or the ‘SIP Proxy’ proposed by the IETF are examples of Call Controllers. If a larger communication network is split into a plurality of domains or ‘zones’, a separate Call Controller can be provided in each domain. It is also possible to operate a domain without a Call Controller. If a plurality of Call Controllers are provided in a domain, only one of these Call Controllers should be activated. From a logical viewpoint, a Call Controller should be considered as separate from the entities. In physical terms, however, it does not have to be implemented in a separate Call Controller entity, but can also be provided in any endpoint of a connection (e.g. designed as an H.323 or SIP endpoint, terminal, media gateway, multipoint control unit), or even in an entity which is primarily designed for program controlled data processing (e.g. computer, PC, server). A physically distributed implementation is also possible.
The Resource Control layer comprises at least one Resource Controller, to which inter alia the following functions are assigned:                Capacity Control: controlling the traffic volume which is supplied via packet streams to the communication network, e.g. by monitoring the transfer capacity of individual packet streams.        Policy Activation (optional): reserving resources in the communication network for transfer of a prioritized packet stream if necessary.        Priority Management (optional): according to the priority of their packet streams, setting and monitoring priority flags in the packets and, if the packets are already flagged with priorities, possibly correcting priority flags in the packets.        
The Resource Controller is also called a ‘Policy Decision Point (PDP)’. It is implemented within so-called Edge Routers, for example, these being known also as Edge Devices, Access Nodes or even Provider Edge Routers (PER) when assigned to an Internet Service Provider (ISP). These Edge Routers can also be designed as Media Gateways to other networks, to which the voice-data networks are connected. These Media Gateways are then connected to both a voice-data network and the other networks, and are used internally for converting between the different protocols of the various networks. The Resource Controller can also be designed solely as a proxy, and forward information that is relevant to the Resource Controller to a separate entity on which the Resource Controller is implemented.
The fundamental interaction between Call Controller and Resource Controller as per the Session Initiation Protocol (SIP) of the IETF or the H.323 protocol family of the ITU is explained using the example of a Call Setup between two VoIP entities which are designed as subscriber terminals. A homogeneous voice-data network is initially taken as a starting point in this case.
As part of the Call Setup, or sometimes even prior to the actual Call Setup, the authentication, authorization and (start of) accounting steps are executed when a terminal dials into the IP network (e.g. via an Internet Service Provider). This so-called ‘AAA’ functionality is usually performed by accessing a subscriber database in which all users are stored, including their identification codes, passwords, permissions, etc. This access is slow and comparatively complex. In the “Best Effort” IP networks of today, this AAA procedure normally takes place once while the user dials in. A further authentication takes place when a Call Controller is used, if the terminal registers at the Call Controller of the Internet Service Provider. According to the ITU standard H.323, this authentication or registration of a terminal at the assigned Gatekeeper is carried out as per the RAS (Registration, Admission, Status) protocol which is described in the ITU standard H.225.0.
The actual Call Setup usually starts in a first step in which the terminals of the subscribers exchange their capabilities (e.g. list of supported CODECs) in order to specify the necessary resources (e.g. bandwidth) and the QoS (e.g. delay, jitter) that is required. The terminals are designed as e.g. IP telephones in the case of voice telephony, and in the case of online video one of the terminals would be designed as a content or application server, e.g. in the network of the ISP.
The exchange of the signaling messages takes place either directly between the devices or via Call Controller switching. In this context, the variant that utilized in the case of each call is individually specified for each terminal and for each transfer direction. The first variant is also called ‘Direct Endpoint Call Signaling’ in the H.323 terminology and the second as ‘Gatekeeper Routed Call Signaling’. In the case of Direct Endpoint Call Signaling, copies of selected signaling messages can be transferred to a Call Controller if necessary, such that a Call Controller is also often aware of the resource and QoS requirements that are agreed between the terminals. However, these requirements are not actively influenced or verified by said Call Controller.
In a second, optional step, the resource and QoS requirement which is agreed in this way can be transferred directly from the terminals of the subscribers to their assigned Resource Controller. After checking the resource and QoS requirement, a confirmation (or rejection) is sent back to the terminal from the Resource Controller.
In a third step which is likewise optional, a policy is activated in the Edge Router and if applicable in other routers in the network, by means of which policy these routers check and ensure that the traffic caused by the terminal is within the limits that were specified in the requirement. An example of such a reservation mechanism is RSVP (resource ReSerVation Protocol).
In order to carry out the three steps a plurality of messages are sent, said messages being used solely for reciprocal agreement among the participating components, and not for transferring the “actual information” between the terminals. This information which is transferred with the messages is usually called signaling information, signaling data, or simply signaling. In this case, the term must be understood in a broad sense. Therefore, for example, the messages as per the RAS protocol, the messages as per the ITU standard H.245 for controlling speech/data channels of existing calls, and all further similarly formed messages are included in addition to the signaling messages. The signaling protocol for the connection setup (call setup) and connection release (call release) according to the ITU is described in the standard H.225.0, for example, and the signaling protocol according to the IETF is described in RFC 2453bis (“SIP: Session Initiation Protocol”). In order to differentiate from the signaling, the “actual information” is also called user information, payload, media information, media data or simply media.
Communication links which are used for transferring the signaling are subsequently called signaling connections. The communication links which are used for transferring the user information are referred to as e.g. voice connection, speech/data channel connection or—more simply—speech/data channel, bearer channel or simply bearer.